The invention relates to PCM-to-PWM signal conversion, and in particular to the correction of non-linearities and noise in the digital conversion of a PCM signal into a PWM signal.
It is known that the comparison of an analog signal amplitude with the amplitude of a higher-frequency reference signal (e.g., a sawtooth waveform) results in a 1-bit digital or pulse width modulated (PWM) signal. This PWM signal can be converted back into an analog signal using a D/A converter. For PWM signals the D/A converter can be a relatively simple lowpass filter.
In this classical PWM circuit, the crosspoint between the analog input signal and the reference sawtooth is used to switch the comparator output between a High (1) and a Low (0). The resulting PWM is called naturally sampled PWM (NPWM). NPWM does not create direct harmonics of the input signal, but only high-frequency inter modulation products between the input and the reference signal, which will be filtered in the D/A process by the lowpass filter.
It has also been known that a successful implementation of a digital PCM-to-PWM conversion needs to mimic the behavior of the above described analog circuit as exact as possible.
Published international patent application No. WO92/11699, incorporated herein by reference, discloses a method of PCM-to-PWM conversion that mimics the above described analog circuit in the digital domain by calculating the crosspoint using a Newton-Raphson Iteration based on a polynomial interpolation of the discrete input signal. Realized as a real time application specifically for class-D amplifiers this method needs a large amount of processing power.
The published international patent application No. WO97/37433, incorporated herein by reference, discloses a method of modeling and therefore predicting the non-linearity of a pulse code modulation to pulse width modulation conversion process. In this model the use of a so-called Hammerstein filter is proposed as a suitable method of correcting several errors of the conversion process. For real-time audio applications (with a bandwidth of 20 Hz . . . 20 kHz), where for practical reasons the PWM pulse repeating frequency is between 200 kHz and 400 kHz, the necessary computing power is still quite large.
A different method has been published by Streitenberger et al. (Zero Position Coding (ZePoC)—A Generalised Concept of Pulse-Length Modulated Signals and its application to Class-D Audio Power Amplifiers, 110th AES Convention, May 12-15, 2001). This paper is incorporated herein by reference. This method uses analytical exponential modulation to generate the PWM signal that results in a lower pulse-repeating frequency (less than 100 kHz for audio applications). The discussed audio implementation needed three DSP engines to process this complex method with a total computing power of 233 MMACs.
It is also known that the non-linearities of the PCM-to-PWM conversion process can be compensated using various feedback techniques combined with adaptive filtering (e.g., TRIPATH White Paper, Apr. 12, 1999). Digital PWM methods are known which calculate the PWM pulse widths from discrete sampling values of the continuous input signal.
Also known is a so-called UPWM (uniform sampling pulse width modulation) method which calculates the pulse widths linearly from (PCM-type) sampling values. However, since the subsequent digital-to-analog conversion of the thus generated PWM signal is a nonlinear process, the entire UPWM method produces large nonlinearities.
German Patent DE 101 56 744 A1 discloses a method in which intermediate values are interpolated for the individual PCM signal values, and in which comparison values are determined in a ramp function relative to the corresponding sampling nodes. An evaluation unit to generate the PWM signal compares the interpolated PCM signal values using the comparison values which follow a ramp pattern. Ultimately, this method too follows the fundamental principle of a method in which an analog or digitized signal amplitude is compared with the amplitude of a higher-frequency reference signal, for example, a saw-tooth signal, in order to generate a pulse-width-modulated signal. Such a signal may be converted back into an analog signal using a digital-to-analog converter. When PWM signals are used, the digital-to-analog converter may consist essentially of a relatively simple low-pass filter. In the classical (analog) PWM circuit, the precise intersection between the input signal and the reference signal is used to connect a comparator in the form of a comparison circuit in order to switch the output signal value between two discrete states. What is generated is the naturally sampled PWM signal (NPWM). After the digital-to-analog conversion, the NPWM signal does not contain any direct harmonic oscillations of the input signal but only higher-frequency inter modulation products between the input signal and the reference signal which are filtered out in the analog-to-digital process by the low-pass filter. Digital PCM-to-PWM conversion circuits, such as that described in DE 101 56 744 A1, must mimic the response of a corresponding analog circuit as precisely as possible. Currently, limitations arise in terms of real-time applications due to the requisite processing time and the pulse width resolution or number of bits. To deal with this, a variety of methods are known, which will be described below.
A method for converting PCM data to PWM data consists in linearly transforming each PCM signal value or PCM sampling value to a corresponding PWM pulse width, as described, for example, by Erik Bresch and Wayne T. Padgett, (“TMS320C67-Based Design of a Digital Audio Power Amplifier Introducing Novel Feedback Strategy”). This transformation involves the so-called uniformly sampled pulse-width modulation (UPWM). However, this method generates a large number of direct harmonic oscillations of the original signal and is thus not suitable for audio applications.
When the PWM sampling period approaches zero, the difference between the pulse widths using UPWM and NPWM decreases to zero for the same input waveform. This is the equivalent of an infinitely large PWM pulse rate.
In order to keep the output low-pass filter simple, the preferred pulse rate is at least 10-20 times the bandwidth of the sampled output signal. For audio applications, the typical bandwidth is 20 kHz, so that the pulse rate employed ranges from 200 kHz to 400 kHz. A higher pulse rate is preferred; however, the efficiency of these power switches beyond 1 MHz declines drastically, thereby reducing the general advantage of a digital switching amplifier. As a result, pulse rates between 200 kHz and 1.5 MHz are preferred in practice.
Another factor is the minimal pulse width. Since in a digital system, only discrete pulse widths are possible, the minimal pulse width determines the resolution, and thus the PWM clock rate. For a PWM pulse rate of 384 kHz and a resolution of 16 bits, that is, CD quality, the PWM clock rate would be 384 kHz×216, or 25.16 GHz. A clock rate this high is obviously not processable in practical terms. In making PWM amplifiers, specifically, for audio applications, a variety of methods have been proposed to overcome these limitations.
A problem with known techniques is that the computational power needed for real time audio applications is still enormous. Except for the ZePoC method this is due to the fact that the PCM-to-PWM conversion has to be done at the final pulse repeating frequency of 200 . . . 400 kHz, which often makes a high order oversampling filter necessary if a primary sampling rate of between 32 kHz and 48 kHz is to be used. In case of the ZePoC method the overall computational power is too high, specifically for so-called single-chip multi-channel solutions.